Voice & Video Services
Build voice and video services for SIP & WebRTC devices with our APIs
Connect your SIP or WebRTC devices to our APIs and offer voice and video communications (VoIP) as part of your service or application. You can also easily allow inbound or outbound voice calls to and from the PSTN network or to a private trunkline.
- SIP Device Management
- Provisioning APIs
- SIP Registrar
- SIP Device Monitoring
- SIP Proxy
- Hosted NAT Traversal
- Hosted SBC
- SIP Authentication
- SIP Encryption
- RTP / Media Encryption
- WebRTC Gateway
- Audio Transcoding
Voice and video services for any application and any device
- No additional infrastructure
- No hardware
- SIP Server functionality included
Connect devices such as
- Telecare devices & wristwatches
- Elevators, intercoms & doorbells
- Cameras & drones
- WebRTC clients
- IP desk- or soft phones
CONNECT & REGISTER
INITIATE A SESSION/CALL
CALL EXTERNALLY
Connect and register SIP devices and applications
The SIP devices are provisioned and registered using native SIP or SIP over secure WebSocket (WebRTC).
Once the SIP devices are registered, they are reachable and ready to initiate or receive SIP sessions, such as voice or video calling.
Initiate SIP sessions from SIP or WebRTC devices
Registered devices can initiate and receive SIP sessions i.e voice calls or video communication between devices.
If SIP devices are behind NAT or a Firewall, a media relay service will be dynamically added to make sure that devices stay reachable as expected.
Initiate voice calling over the PSTN network
Enable external calling by connecting a SIP-trunk to a PSTN provider to allow inbound and outbound PSTN calls.
Connect to a private SIP trunk for your internal SIP service i.e an enterprise PBX or a call center in order to integrate the SIP devices with your existing voice solution.
Documentation
Explore getting started guides and APIs in our Developer Area.
Provision and get started with voice & video services
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